System and method for audio conferencing

ABSTRACT

The present disclosure is directed towards an audio conferencing method. Some embodiments may include receiving, at a first mixing device, an audio signal from a first user associated with an audio conference. Embodiments may further include processing the audio signal at the first mixing device to generate a processed audio signal and transmitting the processed audio signal to a second mixing device, wherein the first mixing device and the second mixing device are distributed over a network in a cascaded configuration. Embodiments may also include receiving, at the second mixing device, a third audio signal from a second user associated with the audio conference and processing the third audio signal at the second mixing device to generate a second processed audio signal.

TECHNICAL FIELD

This disclosure relates to signal processing systems and, moreparticularly, to a system and method for audio conferencing.

BACKGROUND

Existing audio conferencing systems often encounter various problems.Some of these include bandwidth problems, mixer capacity (i.e. inabilityto handle a large number of endpoints), and voice quality issues. Forexample, geographically spread out participant streams have to bebrought to the centralized mixer location before mixing in and the samemixed content may need to be sent back to individual legs. This consumeshigh amounts of bandwidth as the number of participants increases. Also,this may result in numerous voice quality problems related to longdistance transmission (e.g., level imbalances, long echo, etc.).

Moreover, not all participants may be talking at any particular point intime during the audio conference. Existing systems mix unwanted streams,the mixed speech may get clipped, unwanted background noise may getmixed in, etc. This may result in a waste of the processing power on themixer as unwanted streams are mixed, thereby limiting the number ofmaximum participants and/or endpoints in an audio conference.

Further, and with regard to voice quality, background noise fromlisteners and active talkers may be mixed into the audio conferencewhich causes fatigue and prevents the user from focusing on theconversation. Moreover, acoustic echo reflections from hybrid elementsand end points may also make it difficult for a user to follow theconversation. With devices such as mobile phones, desk phones andcomputers as end points there may be unbalanced speech levels fromvarious talkers. The end points may or may not handle all scenarios andthe ones without any enhancement are bound to inject impairments.

SUMMARY OF DISCLOSURE

In one implementation, an audio conferencing method, in accordance withthis disclosure, may include receiving, at a first mixing device, anaudio signal from a first user associated with an audio conference. Themethod may further include processing the audio signal at the firstmixing device to generate a processed audio signal and transmitting theprocessed audio signal to a second mixing device, wherein the firstmixing device and the second mixing device are distributed over anetwork in a cascaded configuration. The method may also includereceiving, at the second mixing device, a third audio signal from asecond user associated with the audio conference and processing thethird audio signal at the second mixing device to generate a secondprocessed audio signal.

One or more of the following features may be included. In someembodiments, the method may include transmitting at least one of thefirst processed audio signal and the second processed audio signal to athird mixing device. The third mixing device and the second mixingdevice may be distributed over a network in a cascaded configuration.The method may further include ranking each user based upon, at least inpart, one or more features associated with each user's audio signal. Insome embodiments, the one or more features may include at least one offrequency of user speech, quality of user speech and duration of userspeech. The method may also include dynamically updating the rankingbased upon, at least in part, the received signals. In some embodiments,processing may include applying a voice quality enhancement algorithm toat least one of the audio signals. In some embodiments, applying a voicequality enhancement algorithm may include applying at least one of noisereduction, echo cancellation, and level control enhancements to at leastone of the audio signals. The method may further include transmitting,from the second mixing device, at least one of a voice intelligibilityand an adaptive level enhancement solution to each user's listeningenvironment. In some embodiments, processing may be based upon, at leastin part, the ranking.

In another implementation, an audio conferencing system is provided. Thesystem may include a first mixing device configured to receive an audiosignal from a first user associated with an audio conference, the firstmixing device further configured to process the audio signal to generatea processed audio signal, the first mixing device further configured totransmit the processed audio signal. The system may further include asecond mixing device configured to receive the processed audio signal,wherein the first mixing device and the second mixing device aredistributed over a network in a cascaded configuration, the secondmixing device configured to receive a third audio signal from a seconduser associated with the audio conference and to process the third audiosignal at the second mixing device to generate a second processed audiosignal.

One or more of the following features may be included. In someembodiments, at least one of the first mixing device and the secondmixing device may be configured to transmit at least one of the firstprocessed audio signal and the second processed audio signal to a thirdmixing device. The third mixing device and the second mixing device maybe distributed over a network in a cascaded configuration. The methodmay further include ranking each user based upon, at least in part, oneor more features associated with each user's audio signal. In someembodiments, the one or more features may include at least one offrequency of user speech, quality of user speech and duration of userspeech. The method may also include dynamically updating the rankingbased upon, at least in part, the received signals. In some embodiments,processing may include applying a voice quality enhancement algorithm toat least one of the audio signals. In some embodiments, applying a voicequality enhancement algorithm may include applying at least one of noisereduction, echo cancellation, and level control enhancements to at leastone of the audio signals. The method may further include transmitting,from the second mixing device, at least one of a voice intelligibilityand an adaptive level enhancement solution to each user's listeningenvironment. In some embodiments, processing may be based upon, at leastin part, the ranking.

The details of one or more implementations are set forth in theaccompanying drawings and the description below. Other features andadvantages will become apparent from the description, the drawings, andthe claims.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a diagrammatic view of an audio conferencing process inaccordance with an embodiment of the present disclosure;

FIG. 2 is a flowchart of an audio conferencing process in accordancewith an embodiment of the present disclosure;

FIG. 3 is a diagrammatic view of a centralized mixer in accordance withan embodiment of the present disclosure;

FIG. 4 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 5 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 6 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 7 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 8 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 9 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 10 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 11 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 12 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 13 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 14 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 15 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 16 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 17 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 18 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 19 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 20 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 21 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure;

FIG. 22 is a diagrammatic view of a system configured to implement anaudio conferencing process in accordance with an embodiment of thepresent disclosure; and

FIG. 23 shows an example of a computer device and a mobile computerdevice that can be used to implement embodiments of the presentdisclosure.

Like reference symbols in the various drawings may indicate likeelements.

DETAILED DESCRIPTION

Embodiments provided herein are directed towards a audio conferencingprocess 10 including an audio conferencing mixer that may be configuredto save computational cycles in mixing the audio streams, and thatreduces bandwidth usage while ensuring speech quality is intact. Audioconferencing process 10 may be configured to address voice quality,bandwidth, and mixer capacity issues by utilizing geographicallycascaded mixers. In some embodiments, audio conferencing process 10 maybe configured to mix only the top-N talkers, which leaves computationalresources on the mixer for other tasks, thus saving transmissionbandwidth. Embodiments may also include performing voice qualityenhancement at all stages of the cascaded mixing system, which may helpto remove background noise, echo, normalize the audio prior to mixing,enhance intelligibility of the mixed stream and to adjust any levels asnecessary.

Referring to FIG. 1, there is shown a audio conferencing process 10 thatmay reside on and may be executed by computer 12, which may be connectedto network 14 (e.g., the Internet or a local area network). Serverapplication 20 may include some or all of the elements of audioconferencing process 10 described herein. Examples of computer 12 mayinclude but are not limited to a single server computer, a series ofserver computers, a single personal computer, a series of personalcomputers, a mini computer, a mainframe computer, an electronic mailserver, a social network server, a text message server, a photo server,a multiprocessor computer, one or more virtual machines running on acomputing cloud, and/or a distributed system. The various components ofcomputer 12 may execute one or more operating systems, examples of whichmay include but are not limited to: Microsoft Windows Server™; NovellNetware™; Redhat Linux™, Unix, or a custom operating system, forexample.

As will be discussed below in greater detail in FIGS. 2-5, audioconferencing process 10 may include receiving (202), at a first mixingdevice, an audio signal from a first user associated with an audioconference. Audio conferencing process 10 may further include processing(204) the audio signal at the first mixing device to generate aprocessed audio signal and transmitting (206) the processed audio signalto a second mixing device, wherein the first mixing device and thesecond mixing device are distributed over a network in a cascadedconfiguration. Audio conferencing process 10 may further includereceiving (208), at the second mixing device, a third audio signal froma second user associated with the audio conference and processing (210)the third audio signal at the second mixing device to generate a secondprocessed audio signal.

The instruction sets and subroutines of audio conferencing process 10,which may be stored on storage device 16 coupled to computer 12, may beexecuted by one or more processors (not shown) and one or more memoryarchitectures (not shown) included within computer 12. Storage device 16may include but is not limited to: a hard disk drive; a flash drive, atape drive; an optical drive; a RAID array; a random access memory(RAM); and a read-only memory (ROM).

Network 14 may be connected to one or more secondary networks (e.g.,network 18), examples of which may include but are not limited to: alocal area network; a wide area network; or an intranet, for example.

In some embodiments, audio conferencing process 10 may reside in wholeor in part on one or more client devices and, as such, may be accessedand/or activated via client applications 22, 24, 26, 28. Examples ofclient applications 22, 24, 26, 28 may include but are not limited to astandard web browser, a customized web browser, or a custom applicationthat can display data to a user. The instruction sets and subroutines ofclient applications 22, 24, 26, 28, which may be stored on storagedevices 30, 32, 34, 36 (respectively) coupled to client electronicdevices 38, 40, 42, 44 (respectively), may be executed by one or moreprocessors (not shown) and one or more memory architectures (not shown)incorporated into client electronic devices 38, 40, 42, 44(respectively).

Storage devices 30, 32, 34, 36 may include but are not limited to: harddisk drives; flash drives, tape drives; optical drives; RAID arrays;random access memories (RAM); and read-only memories (ROM). Examples ofclient electronic devices 38, 40, 42, 44 may include, but are notlimited to, personal computer 38, laptop computer 40, smart phone 42,television 43, notebook computer 44, a server (not shown), adata-enabled, cellular telephone (not shown), and a dedicated networkdevice (not shown).

One or more of client applications 22, 24, 26, 28 may be configured toeffectuate some or all of the functionality of audio conferencingprocess 10. Accordingly, audio conferencing process 10 may be a purelyserver-side application, a purely client-side application, or a hybridserver-side/client-side application that is cooperatively executed byone or more of client applications 22, 24, 26, 28 and audio conferencingprocess 10.

Client electronic devices 38, 40, 42, 44 may each execute an operatingsystem, examples of which may include but are not limited to Apple iOS™,Microsoft Windows™, Android™, Redhat Linux™, or a custom operatingsystem.

Users 46, 48, 50, 52 may access computer 12 and audio conferencingprocess 10 directly through network 14 or through secondary network 18.Further, computer 12 may be connected to network 14 through secondarynetwork 18, as illustrated with phantom link line 54. In someembodiments, users may access audio conferencing process 10 through oneor more telecommunications network facilities 62.

The various client electronic devices may be directly or indirectlycoupled to network 14 (or network 18). For example, personal computer 38is shown directly coupled to network 14 via a hardwired networkconnection. Further, notebook computer 44 is shown directly coupled tonetwork 18 via a hardwired network connection. Laptop computer 40 isshown wirelessly coupled to network 14 via wireless communicationchannel 56 established between laptop computer 40 and wireless accesspoint (i.e., WAP) 58, which is shown directly coupled to network 14. WAP58 may be, for example, an IEEE 802.11a, 802.11b, 802.11g, Wi-Fi, and/orBluetooth device that is capable of establishing wireless communicationchannel 56 between laptop computer 40 and WAP 58. All of the IEEE802.11x specifications may use Ethernet protocol and carrier sensemultiple access with collision avoidance (i.e., CSMA/CA) for pathsharing. The various 802.11x specifications may use phase-shift keying(i.e., PSK) modulation or complementary code keying (i.e., CCK)modulation, for example. Bluetooth is a telecommunications industryspecification that allows e.g., mobile phones, computers, and smartphones to be interconnected using a short-range wireless connection.

Smart phone 42 is shown wirelessly coupled to network 14 via wirelesscommunication channel 60 established between smart phone 42 andtelecommunications network facility 62, which is shown directly coupledto network 14.

Referring now to FIG. 3, an embodiment depicting a centralized mixer 300is shown. In this particular configuration, each participant of theaudio conference may be geographically spread out throughout a givennetwork such as those shown in FIG. 1. Accordingly, each participant'sstream may need to be brought to the centralized mixer location beforemixing in and the same mixed content may need to be sent back to eachindividual leg. This consumes an increasing amount of bandwidth as thenumber of participants increases. Also, this brings with it numerousvoice quality problems related to long distance transmission, some ofwhich may include, but are not limited to level imbalances and longecho.

In some instances, the configuration of FIG. 3 may encounter voicequality issues. For example, background noise from listeners and activetalkers may be mixed into the audio conference, which may detract fromthe conversation. Similarly, acoustic echo reflections from hybridelements and end points makes it difficult to follow the conversation.With different types of devices (e.g. mobile phones, desk phones andcomputers) as end points there may be unbalanced speech levels fromvarious talkers. The end points having voice quality enhancementembedded in them may or may not handle all scenarios and the end pointswithout any enhancement are bound to inject impairments.

Further, in some cases not all of the participants may be talking duringan audio conference. Hence, only those audio streams that have necessaryinformation may need to be mixed. Without this, the mixed speech can getclipped, unwanted background noise can get mixed in and also theprocessing power on the mixer is wasted to mix unwanted streams. Thismay limit the number of maximum participants in an audio conference.

Traditional audio conferencing mixers do not have voice qualityenhancement built in to the system and do not have the capability tocascade. The system of FIG. 3 may be configured to mix the top Nstreams, however, this is based on the instantaneous/average energylevels without knowing if the energy is valid speech or noise or for howlong has it been on the line.

Referring now to FIG. 4, an embodiment of audio conferencing process 10in accordance with the present disclosure is provided. In thisparticular embodiment, audio conferencing process 10 may utilizecascaded mixers distributed geographically as opposed to using just onecentralized mixer as shown in the embodiment of FIG. 3 (e.g. aconference with 10 participants where 4 are in the United States and 6are in Europe with the mixer located in United States—by using one extramixer device placed in Europe only one international link may berequired instead of 6). This type of configuration may help in savingtransmission bandwidth, reducing quality impairments and avoiding theneed to have a powerful centralized machine. Further, audio conferencingprocess 10 may be configured to mix only the top N talkers. This type ofconfiguration may allow for additional computational capacity on themixer for other operations and may help to conserve transmissionbandwidth. The configuration depicted in FIG. 4 is provided merely byway of example as any number of mixers and participants may beassociated with the teachings of the present disclosure.

In some embodiments, audio conferencing process 10 may be configured toperform voice quality enhancement on all the legs at the mixer. As such,audio conferencing process 10 may be configured to perform voice qualityenhancement at all stages of the cascaded mixing system, which may helpto remove background noise, echo, normalize the audio prior to mixing,enhance intelligibility of the mixed stream and to adjust any levels asnecessary.

Embodiments of audio conferencing process 10 described herein may beused with geographically dispersed cascaded mixers in order to handle alarge number of participants. In addition to the particularconfiguration shown in FIG. 4, it should be noted that there may be anynumber of levels of cascading without departing from the scope of thepresent disclosure. For example, although in FIG. 4 the number ofparticipants is limited to 50, embodiments of audio conferencing process10 may be used with any number of participants (e.g. in the order ofthousands). One possible example may include, but is not limited to, anall hands meeting across all of a corporation's global offices. In thiscase particular example, there may be a first level of cascaded mixer atan individual satellite office level. Accordingly, many such offices ina city could be cascaded into a city level mixer—second level. Many suchcities could be cascaded to a country level mixer—third level. Many suchcountries could be cascaded to continent level mixer—fourth level.Additionally and/or alternatively, a fifth level mixer may be used tocentrally mix the legs from each continent. Numerous otherconfigurations may be used as well without departing from the scope ofthe present disclosure.

In some embodiments, audio conferencing process 10 may use one or morevoice activity detector (“VAD”) algorithms (discussed in further detailbelow) in order to accurately detect both speech and non-speech portionsand also to maintain a history of the amount of talking carried out byeach talker. Additional information regarding VAD may be found in UnitedStates Patent Publication Number 2011/0184732 having an application Ser.No. 13/079,705, which is incorporated herein by reference in itsentirety. Additionally and/or alternatively, audio conferencing process10 may utilize noise reduction, echo cancellation and level controlenhancements in conjunction with audio conferencing on the same device.

Embodiments of audio conferencing process 10 may incorporate top Nmixing. As used herein, the phrase “top N mixing” may rank each speakerin the decreasing order of whether to mix his speech or not and may alsodynamically update the rank order. Audio conferencing process 10 mayevaluate whether there is valid speech content from a participant usingone or more voice activity detector algorithms. In some embodiments,this ranking may be based upon a history of how long a person has beentalking. In operation, if a person has been talking for a long time thenthere may be a good chance that he or she will continue to talk so audioconferencing process 10 may rank that leg high for mixing.Alternatively, if audio conferencing process 10 has determined that aparticipant has not spoken for a predetermined and/or extended period oftime then audio conferencing process 10 may rank that participant low.

As discussed above, and referring again to FIG. 4, embodiments of audioconferencing process 10 may utilize cascaded mixing techniques. Forexample, in some embodiments, each cascaded mixer may service a subsetof participants. The top N mixing, discussed above, may be carried outon each of the cascaded mixers and a mixed stream from this cascadedmixer may be generated. This mixed stream may then act as an input tothe next stage of cascade mixer. This process may proceed until reachingthe centralized mixer. The mixed stream of the Top N centralized mixermay be sent down to each of its sub-mixers which plays it back to theparticipants of their respective sub-mixers. In some embodiments, ifnone of the participants of a cascade stage are talking there may not bea mixed stream available from this stage.

Embodiments of audio conferencing process 10 may utilize one or morevoice quality features. Some of these may include, but are not limitedto, noise reduction, echo cancellation and level control enhancements onspeech coming from the participants towards the mixer so that anyimpairments on the ingress can be treated before mixing. Some featuresmay also include voice intelligibility and adaptive level enhancement inthe direction from mixer towards the participant so that individuallistening environments may be addressed. In some embodiments, voicequality enhancement is needed on only those legs that have potential tobe mixed in.

As discussed above, audio conferencing process 10 may utilize adaptivenoise reduction (“ANR”) techniques and one or more voice activitydetector algorithms. In speech communication systems the presence ofbackground interference in the form of additive background and channelnoise may drastically degrade the performance of the system. Embodimentsdisclosed herein may incorporate noise reduction algorithms designed toimprove the performance of communication systems by reducing noise in asingle channel system without introducing audible speech distortion ormusical noise. This type of algorithm may employ advanced spectralsubtraction techniques based on masking properties of the human auditorysystem. The algorithm may continuously restore the natural clean speechagainst a wide variety of noise sources (e.g., car noise, street noise,babble noise, cockpit noise, train noise, harmonic noise, communicationchannel interference, office noise, wind and etc.). Therefore, itdramatically improves the communication quality—both perceptual qualityand signal-to-noise ratio (SNR) measurements.

In some embodiments, ANR operations may include one or more features,some of which may include, but are not limited to, continuously andadaptively removing a wide variety of noise from speech with littlespeech distortions while preserving background noise characteristics.ANR algorithms may include a configurable maximum noise levelsuppression up to 21 dB (21 dB, 18 dB, 15 dB, 12 dB, 9 dB). Although themaximum suppression level of background noise is configurable, theactual level of suppression depends on what the local speech and noisecharacteristics are. For example, if the user configuration is 18 dBmaximum attenuation, but at the situation of current local speech andnoise characteristic, attenuating 18 dB may cause audible artifacts, anANR algorithm may automatically reduce the level of attenuation toprevent naturalness of the original speech. In some embodiments, the ANRalgorithm may include, for example, 15 ms algorithm latency, convergencetime of less than 2 s, approximately 3.87 MCPS processing complexityusing TI TMS320CC54x processor when zero-padding flag is turned on, andapproximately 4.43 MCPS processing complexity using TI TMS320CC54xprocessor when zero-padding flag is turned off. Some embodiments mayutilize an ANR algorithm having a comfort noise floor option withconfigurable noise floor level. Additionally and/or alternatively, anSNR adaptive mode may automatically enable maximum noise reduction forlow-SNR inputs (i.e. SNR<12 dB), while applying moderate or minorreduction to the higher SNR inputs. This may reduce the noiseaggressively when noise is really high, however in less noisyconditions, the level of noise reduction may adapt according to thesignal SNR to minimize the undesirable impact on the speech signal dueto noise reduction processing.

In some embodiments, and to further improve the accuracy of VAD decisionand convergence time for the adaptive noise reduction, four majorimprovements are made to the VAD module. In some embodiments, ahigh-pass filter may be included, which may include (1) the reduction ofnumber of critical band from 18 bands to 17 bands, and (2) someadjustments on boundary mapping of critical bands. In some embodiments,adding a high-pass filter before VAD processing may aid the decision forsome noise type (esp. wind noise). At the same time, other modules inthe system (i.e. tone detection) may need HPF. HPF may be added beforeVAD in some cases. Due to the limited-number of frequency bins, somebins may be mapped into different bands.

Additionally and/or alternatively, some embodiments of the VAD may bedesigned to give bias to active decision since it may be designed aspart of adaptive noise reduction (ANR) module. When the decision is usedfor other purposes (e.g. ALC) the fast recognition of non-active speechbecomes more and more important since this may affect ALC's convergencetime though it may not affect ANR's performance. For example, when theinput is clean on-off high-level tones with very short (e.g., 50 ms)silence gaps, the original VAD may not be able to recognize thesesilence gaps consistently. To overcome this, the improvement made hereis to introduce a short-term energy Es(n) (time constant is about 11ms).

E_(s)^(i)(n) = β E_(s)^(i)(n − 1) + (1 − β)E_(n)^(i)(n), β = 0.1E_(sdB)^(i)(n) = 10log₁₀E_(s)^(i)(n)  dB${\nabla{E_{sdB}(n)}} = {\sum\limits_{{each}\mspace{14mu}{critical}\mspace{14mu}{band}\mspace{14mu} i}\left( {{E_{sdB}^{i}(n)} - {{\overset{\_}{E}}^{i}(n)}} \right)}$

Where i is the index of critical band, n is the index of frame number.When ∇E_(sdB)(n) is below a threshold and the voicing parameter forcurrent frame is low, the current frame is preliminary decided as anon-active frame. Of course, this preliminary decision will be smoothedby VAD hangover later.

Embodiments disclosed herein may improve the VAD initial convergencewhen idle code detection is not available. For example, the original VADassumes the initial 100 ms of input signal are non-active and VAD statesare kept in non-active state. These 100 ms signals are used to build upthe initial VAD state variables, which will affect the initialconvergence rate. This design can aggressively achieve very fast initialconvergence. Embodiments disclosed herein may include both aggressiveoperating mode and normal operating mode for VAD. The aggressiveoperating mode is the same as the original VAD design, when idle codedetection is enabled at “either”, the VAD can be set to this mode tomaintain the faster convergence. While idle code detection is notavailable or provisioned as other options, the VAD should be set asnormal operating mode, in which for the first 40 ms, the VAD statevariables is built up exponentially:

${{Eavg}^{i}(n)} = {{\frac{15}{16}{{Eavg}^{i}\left( {n - 1} \right)}} + {\frac{1}{16}{E^{i}(n)}}}$Eavg^(i)(0) = 0, i = 0, 1, …  , 16

Note n is the frame index, and i is the critical band index.

Embodiments disclosed herein may improve the general VAD convergencetime for various conditions. The following efforts are made to improvethe VAD performance, and consequently to improve the ANR convergencetime: Noise floor power spectral tracking for each critical band andimproved computation of average energy for each critical band.

Noise floor power spectral P^(i)(n) for i-th critical band at frame n istracked even during speech frames. During the speech frame,if P ^(i)(n−1)<E ^(i)(n)P ^(i)(n)=αP ^(i)(n−1)+βE ^(i)(n)+γE ^(i)(n−1),α=0.998,β=0.05,γ=−0.048elseP ^(i)(n)=E ^(i)(n)

Where E^(i)(n) is the input signal power at the i-th critical band atn-th frame.

The time-constant for updating average signal energy in dB ispower-adapted. For the i-th critical band, and for n-th frame:Ē ^(i)(n)=αĒ ^(i)(n−1)+(1−α)E ^(i)(n) in dBWhereα=α_(H)−β(E _(H) −E _(total)),α_(H)=0.97

Embodiments disclosed herein may improve VAD convergence after networkdropouts. As observed from filed captures, GSM switches insert mutepattern when multiple frames are dropped. The long dropouts will resetVAD noise estimation and result in VAD re-convergence after networkrecovery. When the noise level after dropouts is large, it will takequite a long time for VAD to re-recognize the noise frames. The customermay complain the noise coming back after dropouts. To fasten there-convergence time, changes are made in VAD to freeze updating noisespectral contour when such dropouts are detected. This on one hand willspeed up the re-convergence (need no time to converge when noiseunchanged before and after dropouts), on the other hand, this changewill not affect initial convergence time since an initial noise spectralis assumed.

In some embodiments, the ideal noise reduction algorithm will onlyremove noise part from the noisy speech while maintain speech partuntouched. However, in reality this is usually not possible to find suchan ideal algorithm. Therefore the realistic requirement for noisereduction algorithms becomes removing noise as much as possible whilemaintain the speech distortions as low as possible. Spectral shaping isdesigned to work together with ANR algorithm to reduce the perceptualspeech distortion introduced by ANR. The goal of spectral shaping is toreduce perceptible speech artifacts introduced by ANR/NS whilemaintaining ANR/NS's ability to reduce noise. The idea is to boostperceptual important spectral areas of the processed speech, i.e.formants, to maintain the noise reduction the same, the spectral areaswith less perceptual importance are suppressed further. Both objectiveand subjective tests show that spectral shaping improves quality ofANR/NS processed speech, especially for the tandem situation.

In some embodiments, adding a comfort noise floor option may help toavoid quite-line problem found in the field when noise reductionattenuates noise to a level that is too low, people may have dead-lineperception. When the comfort noise floor option is turned on in theconfiguration, a comfort noise floor will be presented in the processedoutput. The noise floor level is configurable, the default value isabout −55 dBov. The option is by default always on when noise silenceris turned on. For ANR feature, this option can be switched through userconfiguration. This option is recommended to be turned off whenconducting any objective testing on ANR. But it is recommended to beturned on when involving subjective listening.

In some embodiments, adding a frame loss handling feature inside VADthrough user-configurable interface of ANC may improve VADre-convergence time when frame loss happened during speech. The designgoals are to maintain fast re-convergence when frame loss happens,especially in the middle of noisy speech and to reduce mis-detection offrame loss which may cause level attenuation for small signal. In orderto achieve the above targets, the following scheme is designed to handlepossible frame loss: Short term frame energy E is computed, if E is verylow and consistent for some period, possible frame loss may happen,within the first 300 ms continuous possible-loss frames, the currentframe will not contribute to the noise spectral updates at all, if thisperiod exceeds 300 ms, only a portion of current frame will contributeto the noise spectral updates, the percentage of contribution will beincreased as time passes. When the possible-loss frames continue morethan 1 s, the current frame will be fully contributed to the noisespectral updates, just like there is no frame loss handling. Theassumption behind this scheme is that most quality-reasonable frame losshappens within 1 s. This feature can be configurable to disable,standard frame loss handling, and high frame loss handling through ANCinterface. The recommended setting is standard. However, if mostenvironments of customers' network are with high noise, while customersstill want to utilize ANC to increase channel capacity by turning onnetwork DTX, high frame loss handling is recommended.

In some embodiments, an SNR-adaptive ANC operation may be employed tobalance the amount of noise reduced and the undesired impact on thespeech signal due to the processing. The design goals of adding theadaptive mode to existing ANC, may include improving tandem ANCsubjective performance, i.e. the cleaner the input is, the lessaggressive attenuation is, and the less artifacts is introduced too. Forexample, if the original input SNR is high enough, the tandem outputwill be very similar to the first ANC output. This will also improveclean speech subjective qualities. The overall SNR-adaptive ANC designis based on the following SNR-Gain function:

${Gain} = \left\{ \begin{matrix}{{- 18}\mspace{14mu}{dB}} & {{{when}\mspace{14mu}{SNR}} \leq {\left( {{\max\;{SNR}} - 18} \right)\mspace{14mu}{dB}}} \\{\left( {{SNR} - {\max\;{SNR}}} \right)\mspace{14mu}{dB}} & {{{when}\mspace{14mu}\left( \;{{\max\;{SNR}} - 18} \right)\mspace{14mu}{dB}} < {SNR} \leq {\max\;{SNR}}} \\{0\mspace{14mu}{dB}} & {{{when}\mspace{14mu}{SNR}} > {\max\;{SNR}}}\end{matrix} \right.$

Referring now to FIG. 5, an embodiment of audio conferencing process 10depicting a master-slave mixing configuration is provided. In thisparticular configuration, a master mixer (“MM”) may be in communicationwith one or more cascading slave mixers (“SMC”), which may each be incommunication with one or more participant slave mixers (“SMP”) throughone or more cascading slave mixers (“SMC. In some embodiments, each SMPmay be configured to connect a certain number of participants. In somecases, one SMC or MM may be connected to this SMP. The SMP may beconfigured to rank the individual participant legs based on talkeractivity. In a first type of configuration (“type-A”), the SMP mayselect the top N streams and send them without mixing to its master ornext cascaded slave mixer. Individual streams from P participantsconnected to this slave mixer may be reduced to N individual streams(e.g., N<P).

In a second type of configuration (“type-B”), the SMP may select the topN streams and send the mixed stream. In this example, the P individualstreams may be reduced to 1. In some embodiments, the SMP may send arank vector towards the next slave/master mixer and may also generatecustomized mixed streams for each individual participant. Each SMP mayalso execute a VQA Application. If Type B, the SMP may apply ANR, AECand/or ALC on individual streams from P participants. If Type A or B,the SMP may apply EVI and ALE to the customized mixed stream to beplayed to each participant.

In some embodiments, each SMC may be connected to a single SMP. In somecases, many SMPs may be connected to a single SMC. For example, oneother SMC in the cascading layer or one MM may be connected to this SMC.Each SMC may be configured to rank the streams (e.g., individual ormixed) and further reduce the set of streams. If Type A, the SMC mayrank the individual streams coming from all SMPs and select the top Nstreams. The SMC may send the individual N streams to the next higherlayer without mixing. If Type B, the SMC may rank the mixed streams andselect the top-X mixed streams. For example, there may be N individualstreams grouped as X mixed streams sent to the next higher layer. Insome embodiments, each SMC may update the rank vector and send ittowards the higher layer. Each SMC may also send mixed stream withoutany modifications coming from the higher layer (e.g. SMC or MM) to thelower layer (e.g. SMC or SMP).

In some embodiments, each master mixer may be connected to a combinationof SMPs and SMCs. Each master mixer may execute one or more VQAApplications. For example, if Type A, the master mixer may apply ANR,AEC and ALC on individual streams entering the master mixer. In someembodiments, the master mixer may be configured to perform ranking andmixing operations. For example, if Type A, the master mixer may beconfigured to rank the individual streams coming in and mix the top-Nparticipants to generate a mixed stream. Alternatively, if Type B, themaster mixer may rank the mixed streams as they enter and mix the top-Nparticipants to generate a mixed stream. The master mixer may beconfigured to send the same mixed stream towards SMCs or SMPs connectedto a master mixer.

Referring now to FIG. 6, an embodiment of audio conferencing process 10depicting a grid mixing configuration is provided. In this particularconfiguration, a generalized version of the peer mixer is shown whereeach grid mixer may be connected to any number of other grid mixers. Asshown in the Figure, redundancy is built in to ensure that the mixedstream from other grid mixers is sent in multiple paths to this gridmixer. Accordingly, even if one path fails there are other redundantpaths available. In some embodiments, Type A and Type B configurationsmay be available. In some embodiments, the redundancy for the grid mixermay result from the fact that the mixed stream/audio content from othermixers may reach this mixer in multiple paths. For example, even if oneor a few paths fails due to transmission errors and/or link failurethere may be information coming from other paths.

Referring now to FIG. 7, an embodiment of audio conferencing process 10depicting a peer mixing configuration is provided. In this particularconfiguration, there may be many participants connected to a particularpeer mixer but at the most only two other peer mixers may be connectedto it. The peer mixing configuration may be configured to perform rankand mix operations as discussed herein. For example, if Type-A, the peermixer may rank and select the top-N talkers from the participantsconnected to it and the individual streams coming from the peer mixersconnected to it. Individual streams may be sent along with a rank vectorto the connected peer mixers. A mixed stream may also be generated. Asused herein, the phrase “top-N” may refer to a subset that may be basedupon, at least in part, an activity ranking.

If Type-B, the peer mixer configuration may be configured to rank andselect the top-N talkers from the participants connected to it as wellas the mixed streams received from the peer mixers connected to it. Amixed stream may be sent along with rank vector to the connected peermixers. A mixed stream may also be generated.

If Type-C, the operation may be similar to that of Type-B except that norank information may be sent to the other peer mixers. For example, ifthere are only two peer mixers connected to each other and there are afew participants connected to each participant.

In some embodiments, a peer mixing configuration may employ one or moreVQA applications. The VQA application may be configured to apply one ormore of ANR, AEC, ALC on individual streams coming from eachparticipant. The VQA application may also apply EVI and ALE oncustomized mixed stream generated for each participant.

Referring now to FIG. 8, an embodiment of audio conferencing process 10depicting a centralized mixer having various VQA components is provided.In this particular example, three participants (i.e., P1-P3) are shown,however, any number of participants may be involved without departingfrom the teachings of the present disclosure. In some embodiments, priorto mixing, noise reduction, echo suppression and operations configuredto normalize levels to the target speech level may be employed. Oncemixing has been performed EVI and ALE may be applied.

Referring now to FIG. 9, an embodiment of audio conferencing process 10depicting a centralized mixer having Top-N mixing and VQA is provided.In this particular embodiments, the voice quality enhancement module(“VM”) may be configured to receive an audio stream from eachparticipant. The VM module may perform ANR followed by AEC and then ALCoperations. In some embodiments, supporting modules, which may include,but are not limited to, VAD and VADPP may also be used to assist in theANR, AEC and ALC operations. In some cases, the VM module may beconfigured to operate only on individual streams. After processing isperformed at the VM module, the stream may be provided directly to themixer or may be sent to a talker activity detector (“TAD”). The TAD maybe configured to determine if a participant has been talking by usingthe VAD decision made by the VAD and other power measurements made inthe VM.

Once the stream is received at the mixer, the streams may be ranked andTop-N mixing may be performed. As discussed above, the ranking vectormay include, but is not limited to, the rank of the participant, activespeech level (e.g., dBm), talker activity (e.g., seconds), how often theparticipant talks in bursts, etc. Accordingly, embodiments of audioconferencing process 10 may be configured to learn the rank of aparticular participant connected to the cascade of mixers connected tothis mixer. Appropriate mixing may then be carried out with thisinformation.

After leaving the mixer, a mixed stream may be generated for eachparticipant using a generate stream module (“GS”). The GS may beconfigured to prevent a participant's voice from being played back tothe same participant. This may be necessary, for example, in order tocreate customized mixed stream for each of the participants. The voicequality enhancement module towards the participant (“VP”) may beconfigured to perform EVI followed by ALE. In some embodiments, the VPmay operate on mixed streams typically and may be the last step beforethe mixed stream is played out to the participant.

Referring now to FIG. 10, an embodiment of audio conferencing process 10depicting a type-A slave mixer for participants (“SMP_(A)”) is provided.As shown in FIG. 10, many of the components and modules described abovewith reference to FIG. 9 may be employed here as well. In someembodiments, the SMP_(A) may be configured to select the top-N talkersfrom the participants. Additionally and/or alternatively, the SMP_(A)may also send individual streams along with the rank vector towards anSMC_(A) and/or an MM_(A). The MM_(A) may mix the top-N individualstreams and may send the mixed stream back to slave mixers. In someembodiments, the SMP_(A) may be configured to generate individual mixedstreams for each of its participants from the mixed stream received fromMM_(A)

Referring now to FIG. 11, an embodiment of audio conferencing process 10depicting a type-A master mixer (“MM_(A)”) is provided. Each MM_(A) maybe configured to send and receive information with one or more SM_(A)mixers. The SM_(A) mixers may be participant and/or cascading in nature.

Referring now to FIG. 12, an embodiment of audio conferencing process 10depicting a type-A slave mixer for cascading (“SMC_(A)”) is provided. Asshown in the Figure, each SMC_(A) may be in communication with one ormore SM_(A) mixers as well as an MM_(A). The SM_(A) mixers may beparticipant and/or cascading in nature.

Referring now to FIG. 13, an embodiment of audio conferencing process 10depicting a type-B slave mixer for participants (“SMP_(B)”) is provided.In this particular embodiment, the SMP_(B) may be configured to selectthe top-N talkers from the participants. A mixed stream may be sentalong with the rank vector towards SMC_(B) or MM_(B). The MM_(B) may beconfigured to mix the top-N individual streams by mixing top-M mixedstreams and transmitting the mixed stream back to the slave mixers. TheSMP_(B) may be configured to generate individual mixed streams for eachof its participants from the mixed stream received from MM_(B).

Referring now to FIG. 14, an embodiment of audio conferencing process 10depicting a type-B master mixer (“MMB”) and/or a type-B slave mixer forcascading (“SMC_(B)”) is provided. In this particular embodiment, theMMB and/or SMC_(B) may be in communication with one or more slave mixers(e.g., SM_(B)(1-3)).

Referring now to FIG. 15, an embodiment of audio conferencing process 10depicting a type-A peer mixer (“PM_(A)”) is provided. In someembodiments, each PM_(A) may be configured to select the top-N talkersfrom the participants. The PM_(A) may also transmit individual streamsalong with the rank vectors to the peer mixers connected to it. Therecould be at most two PM_(A) mixers connected. Each PM_(A) may generate aper participant mixed stream based on the individual streams coming fromother participants and peer mixers connected to it.

Referring now to FIG. 16, an embodiment of audio conferencing process 10depicting a type-B peer mixer (“PM_(B)”) is provided. In someembodiments, each PM_(B) may be configured to select the top-N talkersfrom the participants. The PM_(B) may also transmit the mixed streamalong with the rank vectors to the peer mixers connected to it. Therecould be at most two PM_(B) mixers connected. Each PM_(B) may generate aper participant mixed stream based on the individual streams coming fromother participants and mixed streams coming from other peer mixersconnected to it.

Referring now to FIG. 17, an embodiment of audio conferencing process 10depicting a type-C peer mixer (“PM_(C)”) is provided. The PM_(C) is avariation of the type-B peer mixer PM_(B), where there is no rankingvector shared between mixers. For example, this may be possible wherethere are very few participants connected to a peer mixer and there areonly two peer mixers overall.

Referring now to FIG. 18, an embodiment of audio conferencing process 10depicting a type-A grid mixer (“GM_(A)”) is provided. Each GM_(A) mayfunction similarly to the PM_(A) described above, however, without anyrestriction on the number of mixers connected to it. This type ofconfiguration may allow for additional built-in redundancy. In thefigure, each individual stream is denoted by a solid line and the rankvector is denoted by a dashed line.

Referring now to FIG. 19, an embodiment of audio conferencing process 10depicting a type-B grid mixer (“GM_(B)”) is provided. Like the GM_(A),each GM_(A) may function similarly to the PM_(A) described above,however, without any restriction on the number of mixers connected toit. This type of configuration may allow for additional built-inredundancy. In the figure, each individual stream is denoted by a solidline, the rank vector is denoted by a dashed line, and the top-N mixedstream is denoted by a double line.

Referring now to FIG. 20, an embodiment of audio conferencing process 10depicting an example of top-2 mixing using a type-A master-slave mixingmethod is provided. In this particular embodiment, a master mixer isconnected to three participant slave mixers. Each participant mixer isin communication with a number of participants. In this example, eightparticipants are shown and their speech is labeled accordingly (e.g.,speech from participant P_(i) is denoted by a single line on the figure,while speech from participant P_(i)+participant P_(j) is denoted by adouble line on the figure). Using the teachings of the presentdisclosure, in this particular example the top-2 most active talkers areparticipants 2 and 5.

Referring now to FIG. 21, an embodiment of audio conferencing process 10depicting an example of top-2 mixing using type-A peer mixingmethodology is provided. In this particular embodiment, a peer mixer isconnected to two other peer mixers. Each peer mixer is in communicationwith a number of participants. In this example, ten participants areshown and their speech is labeled accordingly (e.g., speech fromparticipant P_(i) is denoted by a single line on the figure, whilespeech from participant P_(i)+participant P_(j) is denoted by a doubleline on the figure). Using the teachings of the present disclosure, inthis particular example the top-2 most active talkers are participants 2and 5.

Referring now to FIG. 22, an embodiment of audio conferencing process 10depicting an example of top-2 mixing using type-A grid mixingmethodology is provided. In this particular embodiment, a grid mixer isconnected to two other grid mixers. Each grid mixer is in communicationwith a number of participants. In this example, ten participants areshown and their speech is labeled accordingly (e.g., speech fromparticipant P_(i) is denoted by a single line on the figure, whilespeech from participant P_(i)+participant P_(j) is denoted by a doubleline on the figure). Using the teachings of the present disclosure, inthis particular example the top-2 most active talkers are participants 2and 5.

It should be noted that in addition to mixing capabilities, some or allof the mixers described herein may include voice quality enhancementcapabilities. For example, noise reduction (ANR), echo suppression (AEC)and level control (ALC) may be applied on individual streams beforemixing. Enhanced voice intelligibility (EVI) and adaptive levelequalization (ALE) may be applied as the last step on the customizedmixed stream generated for each of the participants.

In some embodiments, only the central mixer may perform mixing, forexample, the other mixers may not mix but transmit the top-N voicestreams instead. This type of arrangement may conserve transmissionbandwidth (e.g., for transmitting N streams instead of a single stream)using discontinuous transmission (“DTX”) and silence descriptor (“SID”).

In some embodiments, each mixer may be configured to send a mixed streamto the central mixer along with the ranked user activity data.Accordingly, the central mixer might accomplish top-N mixing byselecting <N streams. It can accomplish this because each of thosestreams comes with additional information about their correspondingactivity level. In this way, the central mixer may perform a finalcentral ranking of all the users and then select only those streams thatcarry the top-N speakers (e.g., possibly only 1 or 2). This kind ofmixing may also be performed at every mixer level.

In some embodiments, ALE and EVI may occur only on the final outgoinglinks that connect to the users (e.g., not in the mixer). Additionallyand/or alternatively, echo cancellation, noise reduction and levelcontrol may operate at the mixer level, which may also address possiblecoupling between multiple participants across sub-mixers).

In some embodiments, the mixing algorithm on every mixer may be thesame. For example, each mixer may assume that it is receiving inputsfrom multiple other mixers. Inputs from individual users may be treatedas an input from a mixer with one user. If every mixer performs a localranking then the stream from the central mixer may control because thestream and ranking associated with the central mixer may often have thehighest activity level.

Traditional audio conferencing mixers do not offer certain features fora participant dialing in from a telephone or a mobile phone. Embodimentsdisclosed herein may enable a participant dialing in using traditionaltelephone or mobile phone to use certain productivity tools, some ofwhich are easily possible in a web portal based audio conferencing butneeds signal processing approach to provide them for telephone calls.Accordingly, the mixer may include the capability to understand thesequence of DTMF digits punched in by any participant in-band orout-of-band of speech media.

Embodiments of audio conferencing process 10 may also enable aparticipant in an audio conference to join using a traditional telephoneor mobile phone. Accordingly, a user may be allowed to catch up with thelast “N” seconds of the conference. For example, a participant couldenter a predefined dual-tone multi-frequency (“DTMF”) sequencecombination and listen to the last N seconds of conversation that he/shemight have missed as a result of stepping out of the conference orjoining late. In some embodiments, the process may include buffering upthe mixed stream generated by mixing the voice streams from allparticipants in an audio conference. Depending on the memory availableon the machine in which the audio conferencing is running, the value ofN may be defined. The process may also include playing back the last Nseconds of mixed stream on top of the current running audio conference.In some embodiments, upon entering the predefined DTMF sequence themixer may be notified that this specific participant is requesting theplayback and the mixer may generate a new mixed stream for thisparticipant where the mixed stream from the ongoing conference is mixedwith the recorded mixed stream. The ongoing mixed stream could be mixedbelow the recorded mixed stream (e.g. 6-9 dB) so that it is played inthe background. After the playback is completed the mixer may send theongoing mixed stream at its original level.

Additionally and/or alternatively, in some embodiments a mute indicatormay be provided. In this example, a participant could enter a predefinedDTMF sequence combination and listen to an announcement indicating ifhe/she is on mute or not. Although traditional conferences offer apredefined DTMF sequence to enable or disable mute, most participantswould prefer using the mute option on their handsets as each conferencecould have a separate predefined mute enable sequence. In this example,a participant may enter a predefined DTMF sequence to obtain the statusof the mute. The mixer may measure the energy coming on the leg fromthis participant and if it is less than a certain threshold (e.g., −45dBm) then the mixer may send a mute notification as an announcement tothe participant. The mixer may also use the information of the state ofthat participant if he has enabled or disabled mute on that line. Thisapproach still works if the participant uses handset or the conferenceto mute the line.

In some embodiments, a virtual hand raising option may also be provided.For example, in an online classroom where one speaker (e.g. teacher) islecturing while majority of the participants (e.g. students) arelistening, a participant can raise his/her hand virtually to ask aquestion by entering a predefined DTMF sequence. The host (e.g. teacher)in this case may hear a special interrupt tone to which the teacher canact at a meaningful stop. The mixer may act upon receiving the DTMFsequence and may play a notification tone to the moderator. Themoderator may then interrupt the conversation at a convenient time totake the question from this participant.

In some embodiments, a list of participants may also be provided. Inthis example, a participant could enter a predefined DTMF sequencecombination and listen to a list of participants on the conferencewithout interrupting the conversation. Upon receiving the sequence, themixer may generate a new audio notification containing a list ofparticipants on the call. Concatenating the recording of the name of theparticipant as each participant joins the conference can generate thisaudio of list of participants. The mixer can mix this audio of list ofparticipants on top of the ongoing conference at a level (e.g., 6-9 dB)lower than the ongoing conference.

Referring now to FIG. 23, an example of a generic computer device 2300and a generic mobile computer device 550, which may be used with thetechniques described herein is provided. Computing device 2300 isintended to represent various forms of digital computers, such as tabletcomputers, laptops, desktops, workstations, personal digital assistants,servers, blade servers, mainframes, and other appropriate computers. Insome embodiments, computing device 550 can include various forms ofmobile devices, such as personal digital assistants, cellulartelephones, smartphones, and other similar computing devices. Computingdevice 550 and/or computing device 2300 may also include other devices,such as televisions with one or more processors embedded therein orattached thereto. The components shown here, their connections andrelationships, and their functions, are meant to be exemplary only, andare not meant to limit implementations of the inventions describedand/or claimed in this document.

In some embodiments, computing device 2300 may include processor 502,memory 504, a storage device 506, a high-speed interface 508 connectingto memory 504 and high-speed expansion ports 510, and a low speedinterface 512 connecting to low speed bus 514 and storage device 506.Each of the components 502, 504, 506, 508, 510, and 512, may beinterconnected using various busses, and may be mounted on a commonmotherboard or in other manners as appropriate. The processor 502 canprocess instructions for execution within the computing device 2300,including instructions stored in the memory 504 or on the storage device506 to display graphical information for a GUI on an externalinput/output device, such as display 516 coupled to high speed interface508. In other implementations, multiple processors and/or multiple busesmay be used, as appropriate, along with multiple memories and types ofmemory. Also, multiple computing devices 2300 may be connected, witheach device providing portions of the necessary operations (e.g., as aserver bank, a group of blade servers, or a multiprocessor system).

Memory 504 may store information within the computing device 2300. Inone implementation, the memory 504 may be a volatile memory unit orunits. In another implementation, the memory 504 may be a non-volatilememory unit or units. The memory 504 may also be another form ofcomputer-readable medium, such as a magnetic or optical disk.

Storage device 506 may be capable of providing mass storage for thecomputing device 2300. In one implementation, the storage device 506 maybe or contain a computer-readable medium, such as a floppy disk device,a hard disk device, an optical disk device, or a tape device, a flashmemory or other similar solid state memory device, or an array ofdevices, including devices in a storage area network or otherconfigurations. A computer program product can be tangibly embodied inan information carrier. The computer program product may also containinstructions that, when executed, perform one or more methods, such asthose described above. The information carrier is a computer- ormachine-readable medium, such as the memory 504, the storage device 506,memory on processor 502, or a propagated signal.

High speed controller 508 may manage bandwidth-intensive operations forthe computing device 2300, while the low speed controller 512 may managelower bandwidth-intensive operations. Such allocation of functions isexemplary only. In one implementation, the high-speed controller 508 maybe coupled to memory 504, display 516 (e.g., through a graphicsprocessor or accelerator), and to high-speed expansion ports 510, whichmay accept various expansion cards (not shown). In the implementation,low-speed controller 512 is coupled to storage device 506 and low-speedexpansion port 514. The low-speed expansion port, which may includevarious communication ports (e.g., USB, Bluetooth, Ethernet, wirelessEthernet) may be coupled to one or more input/output devices, such as akeyboard, a pointing device, a scanner, or a networking device such as aswitch or router, e.g., through a network adapter.

Computing device 2300 may be implemented in a number of different forms,as shown in the figure. For example, it may be implemented as a standardserver 520, or multiple times in a group of such servers. It may also beimplemented as part of a rack server system 524. In addition, it may beimplemented in a personal computer such as a laptop computer 522.Alternatively, components from computing device 2300 may be combinedwith other components in a mobile device (not shown), such as device550. Each of such devices may contain one or more of computing device2300, 550, and an entire system may be made up of multiple computingdevices 2300, 550 communicating with each other.

Computing device 550 may include a processor 552, memory 564, aninput/output device such as a display 554, a communication interface566, and a transceiver 568, among other components. The device 550 mayalso be provided with a storage device, such as a microdrive or otherdevice, to provide additional storage. Each of the components 550, 552,564, 554, 566, and 568, may be interconnected using various buses, andseveral of the components may be mounted on a common motherboard or inother manners as appropriate.

Processor 552 may execute instructions within the computing device 550,including instructions stored in the memory 564. The processor may beimplemented as a chipset of chips that include separate and multipleanalog and digital processors. The processor may provide, for example,for coordination of the other components of the device 550, such ascontrol of user interfaces, applications run by device 550, and wirelesscommunication by device 550.

In some embodiments, processor 552 may communicate with a user throughcontrol interface 558 and display interface 556 coupled to a display554. The display 554 may be, for example, a TFT LCD(Thin-Film-Transistor Liquid Crystal Display) or an OLED (Organic LightEmitting Diode) display, or other appropriate display technology. Thedisplay interface 556 may comprise appropriate circuitry for driving thedisplay 554 to present graphical and other information to a user. Thecontrol interface 558 may receive commands from a user and convert themfor submission to the processor 552. In addition, an external interface562 may be provide in communication with processor 552, so as to enablenear area communication of device 550 with other devices. Externalinterface 562 may provide, for example, for wired communication in someimplementations, or for wireless communication in other implementations,and multiple interfaces may also be used.

In some embodiments, memory 564 may store information within thecomputing device 550. The memory 564 can be implemented as one or moreof a computer-readable medium or media, a volatile memory unit or units,or a non-volatile memory unit or units. Expansion memory 574 may also beprovided and connected to device 550 through expansion interface 572,which may include, for example, a SIMM (Single In Line Memory Module)card interface. Such expansion memory 574 may provide extra storagespace for device 550, or may also store applications or otherinformation for device 550. Specifically, expansion memory 574 mayinclude instructions to carry out or supplement the processes describedabove, and may include secure information also. Thus, for example,expansion memory 574 may be provide as a security module for device 550,and may be programmed with instructions that permit secure use of device550. In addition, secure applications may be provided via the SIMMcards, along with additional information, such as placing identifyinginformation on the SIMM card in a non-hackable manner.

The memory may include, for example, flash memory and/or NVRAM memory,as discussed below. In one implementation, a computer program product istangibly embodied in an information carrier. The computer programproduct may contain instructions that, when executed, perform one ormore methods, such as those described above. The information carrier maybe a computer- or machine-readable medium, such as the memory 564,expansion memory 574, memory on processor 552, or a propagated signalthat may be received, for example, over transceiver 568 or externalinterface 562.

Device 550 may communicate wirelessly through communication interface566, which may include digital signal processing circuitry wherenecessary. Communication interface 566 may provide for communicationsunder various modes or protocols, such as GSM voice calls, SMS, EMS, orMMS speech recognition, CDMA, TDMA, PDC, WCDMA, CDMA2000, or GPRS, amongothers. Such communication may occur, for example, throughradio-frequency transceiver 568. In addition, short-range communicationmay occur, such as using a Bluetooth, WiFi, or other such transceiver(not shown). In addition, GPS (Global Positioning System) receivermodule 570 may provide additional navigation- and location-relatedwireless data to device 550, which may be used as appropriate byapplications running on device 550.

Device 550 may also communicate audibly using audio codec 560, which mayreceive spoken information from a user and convert it to usable digitalinformation. Audio codec 560 may likewise generate audible sound for auser, such as through a speaker, e.g., in a handset of device 550. Suchsound may include sound from voice telephone calls, may include recordedsound (e.g., voice messages, music files, etc.) and may also includesound generated by applications operating on device 550.

Computing device 550 may be implemented in a number of different forms,as shown in the figure. For example, it may be implemented as a cellulartelephone 580. It may also be implemented as part of a smartphone 582,personal digital assistant, remote control, or other similar mobiledevice.

Various implementations of the systems and techniques described here canbe realized in digital electronic circuitry, integrated circuitry,specially designed ASICs (application specific integrated circuits),computer hardware, firmware, software, and/or combinations thereof.These various implementations can include implementation in one or morecomputer programs that are executable and/or interpretable on aprogrammable system including at least one programmable processor, whichmay be special or general purpose, coupled to receive data andinstructions from, and to transmit data and instructions to, a storagesystem, at least one input device, and at least one output device.

These computer programs (also known as programs, software, softwareapplications or code) include machine instructions for a programmableprocessor, and can be implemented in a high-level procedural and/orobject-oriented programming language, and/or in assembly/machinelanguage. As used herein, the terms “machine-readable medium”“computer-readable medium” refers to any computer program product,apparatus and/or device (e.g., magnetic discs, optical disks, memory,Programmable Logic Devices (PLDs)) used to provide machine instructionsand/or data to a programmable processor, including a machine-readablemedium that receives machine instructions as a machine-readable signal.The term “machine-readable signal” refers to any signal used to providemachine instructions and/or data to a programmable processor.

As will be appreciated by one skilled in the art, the present disclosuremay be embodied as a method, system, or computer program product.Accordingly, the present disclosure may take the form of an entirelyhardware embodiment, an entirely software embodiment (includingfirmware, resident software, micro-code, etc.) or an embodimentcombining software and hardware aspects that may all generally bereferred to herein as a “circuit,” “module” or “system.” Furthermore,the present disclosure may take the form of a computer program producton a computer-usable storage medium having computer-usable program codeembodied in the medium.

Any suitable computer usable or computer readable medium may beutilized. The computer-usable or computer-readable medium may be, forexample but not limited to, an electronic, magnetic, optical,electromagnetic, infrared, or semiconductor system, apparatus, device,or propagation medium. More specific examples (a non-exhaustive list) ofthe computer-readable medium would include the following: an electricalconnection having one or more wires, a portable computer diskette, ahard disk, a random access memory (RAM), a read-only memory (ROM), anerasable programmable read-only memory (EPROM or Flash memory), anoptical fiber, a portable compact disc read-only memory (CD-ROM), anoptical storage device, a transmission media such as those supportingthe Internet or an intranet, or a magnetic storage device. Note that thecomputer-usable or computer-readable medium could even be paper oranother suitable medium upon which the program is printed, as theprogram can be electronically captured, via, for instance, opticalscanning of the paper or other medium, then compiled, interpreted, orotherwise processed in a suitable manner, if necessary, and then storedin a computer memory. In the context of this document, a computer-usableor computer-readable medium may be any medium that can contain, store,communicate, propagate, or transport the program for use by or inconnection with the instruction execution system, apparatus, or device.

Computer program code for carrying out operations of the presentdisclosure may be written in an object oriented programming languagesuch as Java, Smalltalk, C++ or the like. However, the computer programcode for carrying out operations of the present disclosure may also bewritten in conventional procedural programming languages, such as the“C” programming language or similar programming languages. The programcode may execute entirely on the user's computer, partly on the user'scomputer, as a stand-alone software package, partly on the user'scomputer and partly on a remote computer or entirely on the remotecomputer or server. In the latter scenario, the remote computer may beconnected to the user's computer through a local area network (LAN) or awide area network (WAN), or the connection may be made to an externalcomputer (for example, through the Internet using an Internet ServiceProvider).

The present disclosure is described below with reference to flowchartillustrations and/or block diagrams of methods, apparatus (systems) andcomputer program products according to embodiments of the disclosure. Itwill be understood that each block of the flowchart illustrations and/orblock diagrams, and combinations of blocks in the flowchartillustrations and/or block diagrams, can be implemented by computerprogram instructions. These computer program instructions may beprovided to a processor of a general purpose computer, special purposecomputer, or other programmable data processing apparatus to produce amachine, such that the instructions, which execute via the processor ofthe computer or other programmable data processing apparatus, createmeans for implementing the functions/acts specified in the flowchartand/or block diagram block or blocks.

These computer program instructions may also be stored in acomputer-readable memory that can direct a computer or otherprogrammable data processing apparatus to function in a particularmanner, such that the instructions stored in the computer-readablememory produce an article of manufacture including instruction meanswhich implement the function/act specified in the flowchart and/or blockdiagram block or blocks.

The computer program instructions may also be loaded onto a computer orother programmable data processing apparatus to cause a series ofoperational steps to be performed on the computer or other programmableapparatus to produce a computer implemented process such that theinstructions which execute on the computer or other programmableapparatus provide steps for implementing the functions/acts specified inthe flowchart and/or block diagram block or blocks.

To provide for interaction with a user, the systems and techniquesdescribed here can be implemented on a computer having a display device(e.g., a CRT (cathode ray tube) or LCD (liquid crystal display) monitor)for displaying information to the user and a keyboard and a pointingdevice (e.g., a mouse or a trackball) by which the user can provideinput to the computer. Other kinds of devices can be used to provide forinteraction with a user as well; for example, feedback provided to theuser can be any form of sensory feedback (e.g., visual feedback,auditory feedback, or tactile feedback); and input from the user can bereceived in any form, including acoustic, speech, or tactile input.

The systems and techniques described here may be implemented in acomputing system that includes a back end component (e.g., as a dataserver), or that includes a middleware component (e.g., an applicationserver), or that includes a front end component (e.g., a client computerhaving a graphical user interface or a Web browser through which a usercan interact with an implementation of the systems and techniquesdescribed here), or any combination of such back end, middleware, orfront end components. The components of the system can be interconnectedby any form or medium of digital data communication (e.g., acommunication network). Examples of communication networks include alocal area network (“LAN”), a wide area network (“WAN”), and theInternet.

The computing system may include clients and servers. A client andserver are generally remote from each other and typically interactthrough a communication network. The relationship of client and serverarises by virtue of computer programs running on the respectivecomputers and having a client-server relationship to each other.

The flowchart and block diagrams in the figures illustrate thearchitecture, functionality, and operation of possible implementationsof systems, methods and computer program products according to variousembodiments of the present disclosure. In this regard, each block in theflowchart or block diagrams may represent a module, segment, or portionof code, which comprises one or more executable instructions forimplementing the specified logical function(s). It should also be notedthat, in some alternative implementations, the functions noted in theblock may occur out of the order noted in the figures. For example, twoblocks shown in succession may, in fact, be executed substantiallyconcurrently, or the blocks may sometimes be executed in the reverseorder, depending upon the functionality involved. It will also be notedthat each block of the block diagrams and/or flowchart illustration, andcombinations of blocks in the block diagrams and/or flowchartillustration, can be implemented by special purpose hardware-basedsystems that perform the specified functions or acts, or combinations ofspecial purpose hardware and computer instructions.

The terminology used herein is for the purpose of describing particularembodiments only and is not intended to be limiting of the disclosure.As used herein, the singular forms “a”, “an” and “the” are intended toinclude the plural forms as well, unless the context clearly indicatesotherwise. It will be further understood that the terms “comprises”and/or “comprising,” when used in this specification, specify thepresence of stated features, integers, steps, operations, elements,and/or components, but do not preclude the presence or addition of oneor more other features, integers, steps, operations, elements,components, and/or groups thereof.

The corresponding structures, materials, acts, and equivalents of allmeans or step plus function elements in the claims below are intended toinclude any structure, material, or act for performing the function incombination with other claimed elements as specifically claimed. Thedescription of the present disclosure has been presented for purposes ofillustration and description, but is not intended to be exhaustive orlimited to the disclosure in the form disclosed. Many modifications andvariations will be apparent to those of ordinary skill in the artwithout departing from the scope and spirit of the disclosure. Theembodiment was chosen and described in order to best explain theprinciples of the disclosure and the practical application, and toenable others of ordinary skill in the art to understand the disclosurefor various embodiments with various modifications as are suited to theparticular use contemplated.

Having thus described the disclosure of the present application indetail and by reference to embodiments thereof, it will be apparent thatmodifications and variations are possible without departing from thescope of the disclosure defined in the appended claims.

What is claimed is:
 1. A computer-implemented method for audioconferencing comprising: receiving, at a first mixing device, a firstset of audio signals including a first audio signal associated with anaudio conference; generating a ranking, at the first mixing device, ofeach user associated with the first set of audio signals based upon oneor more features of the first set of audio signals including a speakinglength of time of each user associated with the first set of audiosignals; processing at least one audio signal of the first set of audiosignals at the first mixing device to generate a first set of processedaudio signals based upon, at least in part, the ranking of each userassociated with the first set of audio signals, wherein processing theat least one audio signal of the first set of audio signals at the firstmixing device includes applying a voice quality enhancement algorithm tothe at least one audio signal of the first set of audio signals;transmitting at least one audio signal of the first set of processedaudio signals to a second mixing device, wherein the first mixing deviceand the second mixing device are distributed over a network in acascaded configuration; receiving, at the second mixing device, a secondset of audio signals including a second audio signal associated with theaudio conference; generating a ranking, at the second mixing device, ofeach user associated with the second set of audio signals based upon oneor more features of the second set of audio signals including a speakinglength of time of each user associated with the second set of audiosignals; dynamically updating at least one of the rankings based upon,at least in part, at least one of the first audio signal and the secondaudio signal; and processing at least one audio signal of the second setof audio signals at the second mixing device to generate a second set ofprocessed audio signals based upon, at least in part, the ranking ofeach user associated with the second set of audio signals, whereinprocessing the at least one audio signal of the second set of audiosignals at the second mixing device includes applying the voice qualityenhancement algorithm to the at least one audio signal of the second setof audio signals, wherein applying the voice quality enhancementalgorithm includes applying at least one of noise reduction, echocancellation, and level control enhancements to at least one of thefirst audio signal and the second audio signal.
 2. The method of claim1, further comprising: transmitting an audio signal of at least one ofthe first set of processed audio signals and the second set of processedaudio signals to a third mixing device.
 3. The method of claim 2,wherein the third mixing device and the second mixing device aredistributed over the network in the cascaded configuration.
 4. Themethod of claim 1, wherein the one or more features further include atleast one of frequency of user speech, quality of user speech andduration of user speech.
 5. The method of claim 1, further comprising:transmitting, from the second mixing device, at least one of a voiceintelligibility and an adaptive level enhancement solution to alistening environment of each user.
 6. An audio conferencing systemcomprising: a first mixing device configured to receive a first set ofaudio signals including a first audio signal associated with an audioconference, the first mixing device further configured to generate aranking of each user associated with the first set of audio signalsbased upon, at least in part, one or more features of the first set ofaudio signals including a speaking length of time of each userassociated with the first set of audio signals, the first mixing devicefurther to process at least one audio signal of the first set of audiosignals at the first mixing device to generate a first set of processedaudio signals based upon, at least in part, the ranking of each userassociated with the first set of audio signals, wherein processing theat least one audio signal of the first set of audio signals at the firstmixing device includes applying the voice quality enhancement algorithmto the at least one audio signal of the first set of audio signals, thefirst mixing device further to transmit at least one audio signal of thefirst set of processed audio signals to a second mixing device; and asecond mixing device configured to receive the first processed audiosignal, wherein the first mixing device and the second mixing device aredistributed over a network in a cascaded configuration, the secondmixing device configured to receive a second set of audio signalsincluding a second audio signal associated with the audio conference,the second mixing device further configured to generate a ranking ofeach user associated with the second set of audio signals based upon, atleast in part, one or more features of the second set of audio signalsincluding a speaking length of time of each user associated with thesecond set of audio signals, the second mixing device further configuredto dynamically update at least one of the rankings based upon, at leastin part, at least one of the first audio signal and the second audiosignal, the second mixing device further to process at least one audiosignal of the second set of audio signals at the second mixing device togenerate a second set of processed audio signals based upon, at least inpart, the ranking of each user associated with the second set of audiosignals, wherein processing the at least one audio signal of the secondset of audio signals at the second mixing device includes applying thevoice quality enhancement algorithm to the at least one audio signal ofthe second set of audio signals, wherein applying the voice qualityenhancement algorithm includes applying at least one of noise reduction,echo cancellation, and level control enhancements to at least one of thefirst audio signal and the second audio signal.
 7. The system of claim6, wherein at least one of the first mixing device and the second mixingdevice are further configured to transmit an audio signal of at leastone of the first set of processed audio signals and the second set ofprocessed audio signals to a third mixing device.
 8. The system of claim7, wherein the third mixing device and the second mixing device aredistributed over the network in the cascaded configuration.
 9. Thesystem of claim 6, wherein the one or more further include at least oneof frequency of user speech, quality of user speech and duration of userspeech.
 10. The system of claim 6, wherein the second mixing device isfurther configured to transmit at least one of a voice intelligibilityand an adaptive level enhancement solution to a listening environment ofeach user.
 11. A computer program product residing on a non-transitorycomputer readable storage medium having a plurality of instructionsstored thereon which, when executed across one or more processors,causes at least a portion of the one or more processors to performoperations comprising: receiving, at a first mixing device, a first setof audio signals including a first audio signal associated with an audioconference; generating a ranking, at the first mixing device, of eachuser associated with the first set of audio signals based upon one ormore features of the first set of audio signals including a speakinglength of time of each user associated with the first set of audiosignals; processing at least one audio signal of the first set of audiosignals at the first mixing device to generate a first set of processedaudio signals based upon, at least in part, the ranking of each userassociated with the first set of audio signals, wherein processing theat least one audio signal of the first set of audio signals at the firstmixing device includes applying a voice quality enhancement algorithm tothe at least one audio signal of the first set of audio signals;transmitting at least one audio signal of the first set of processedaudio signals to a second mixing device, wherein the first mixing deviceand the second mixing device are distributed over a network in acascaded configuration; receiving, at the second mixing device, a secondset of audio signals including a second audio signal associated with theaudio conference; generating a ranking, at the second mixing device, ofeach user associated with the second set of audio signals based upon oneor more features of the second set of audio signals including a speakinglength of time of each user associated with the second set of audiosignals; dynamically updating at least one of the rankings based upon,at least in part, at least one of the first audio signal and the secondaudio signal; and processing at least one audio signal of the second setof audio signals at the second mixing device to generate a second set ofprocessed audio signals based upon, at least in part, the ranking ofeach user associated with the second set of audio signals, whereinprocessing the at least one audio signal of the second set of audiosignals at the second mixing device includes applying the voice qualityenhancement algorithm to the at least one audio signal of the second setof audio signals, wherein applying the voice quality enhancementalgorithm includes applying at least one of noise reduction, echocancellation, and level control enhancements to at least one of thefirst audio signal and the second audio signal.